(1) Field of Invention
This invention relates to methods and systems, implementing the method, for audio switching and conferencing.
The invention further relates to a device implementing the method.
The invention further relates to a computer program, which is stored on a computer readable storage media, and which is suitable to perform the method according to the first paragraph when it is run on a digital computer.
The inventions further relates to a computer program product directly loadable into the memory of a programmable device, comprising software code portions for performing the steps of a method according to the first paragraph when said product is run on the device.
(2) Prior Art
Ordinary ‘in-house telephones’ are still commonly used for domestic or corporate purposes. To increase the user-friendliness, these telephones are often cordless telephones, for example according to the Digital Enhanced Cordless Telecommunications ETSI-standard for digital cordless telephones. Such cordless telephone systems comprise a base station and at least one portable handset that is arranged to communicate with the base station, for example via a wireless communication link. FIG. 1 schematically shows a conventional cordless telephone system comprising a base station BS and three portable handsets HS1, HS2, HS3. The handsets HS1,HS2,HS3 comprise a processing unit P1, and have a microphone M1 and a speaker S1 associated with it. The microphone M1 and the speaker S1 may be comprised by the handset HS1, HS2, HS3, but may also one or two remote components arranged to communicate with the processing unit P1 via a wireless or wired communication link (for instance using Bluetooth).
The microphone M1 is arranged to detect sound and transmit a signal representing the detected sound to the processing unit P1. The processing unit P1 is arranged to transmit this signal to the base station BS, for instance via a wireless communication link. The processing arrangement P1 is further arranged to receive a signal from the base station BS and transmit this signal to the speaker S1. The speaker S1 is arranged to generate a sound signal based on the received signal. The handsets HS1,HS2,HS3 may be arranged to carry out all kinds of suitable signal processing steps, such as analogue-to-digital conversion, digital-to-analogue conversion, filtering to increase the quality of the signal, etc. The base station BS may comprise all kinds of hardware and/or software components arranged to receive and transmit signals representing sound from and to the handsets HS1,HS2,HS3. The base station BS is further arranged to communicate with a suitable network NW, e.g. via a line interface LIF, for instance interfacing network NW, which may be the public switched telephone network (PSTN), (asymmetric) digital subscriber line ((A)DSL), integrated services digital network (ISDN). The voice carrier protocol used over these networks may be Voice over IP (VoIP).
A conventional method for audio switching and conferencing for use in such base station exchanges voice data with the external environment via one or more audio channels. An example is voice data exchanged over a public subscriber transmission (PSTN) line, or voice data transmitted through the air for a wireless phone device. Audio Channels are usually bidirectional, for example a PSTN line or unidirectional for example for connecting a speaker or microphone to an audio channel. The bidirectional audio channel has both an input and an output audio stream. The unidirectional audio channel has only one of the two audio streams. Audio channels transfer voice audio data either in raw format or in compressed format. Voice Audio data of the input audio stream can only be processed in a PCM format.
A single bidirectional Audio Channel may correspond to an active conversation between two parties. For operating a bidirectional audio channel the conventional method comprises    capturing Audio data from Party HS1,    converting audio data to PCM format data,    processing the PCM format data,    converting the processed PCM format data to an output compression codec and transmitting the compressed data to party HS2.
Data from Party HS2 follow a similar, but opposite route to Party HS1. The conventional method can be extended with a third party HS3 in order to support audio conferencing. In order to host a three-way audio conference for party HS1, HS2 and HS3 the base station is arranged such that party HS1 listens to the added voice of party HS1 and party HS3, party HS2 listens to the added voice of party HS1 and party HS3, and party HS3 listens to the added voice of party HS1 and party HS2. Thereto, a control unit of the base station is adapted to maintain a list of calls and conferences. This list contains multiple entries each containing a participating channel. A digital signal processor of the base station deals with one item of the list at a time. In order to add a new party to the conference or to set up a new connection or conference, the control unit has to extend or refresh the list and to communicate corresponding changes to the digital signal processor. The digital signal processor has to process all the different combinations of the parties of the participating channels and thus the complexity of the computations increases with an increasing number of parties. Furthermore, special functions as for example microphone or eavesdropping has to be implemented separately and increase the complexity even more.
The conventional method may be implemented in a base station which has limited availability of computational resources and which may comprise a multi layer or multi-processor architecture, wherein the voice processing block is implemented in a separate processor or software block.
The conventional method may become increasingly complex as more than three parties are connected and simultaneous conferences between groups of the connected parties have to be established.